Relay-Version: version B 2.10 5/3/83; site utzoo.UUCP Posting-Version: version B 2.10.2 9/18/84; site amdimage.UUCP Path: utzoo!linus!philabs!prls!amdimage!steve From: steve@amdimage.UUCP (Steve eidson) Newsgroups: net.audio Subject: one real CD interpolation filter Message-ID: <519@amdimage.UUCP> Date: Thu, 29-Aug-85 12:32:59 EDT Article-I.D.: amdimage.519 Posted: Thu Aug 29 12:32:59 1985 Date-Received: Sun, 1-Sep-85 08:44:36 EDT Distribution: net Organization: AMDIMAGE, Sunnyvale, CA Lines: 59 I'm not trying to belabor a point, but I thought I'd post a follow-up now that I have some information on real CD interplation filters. (At least before someone tries to beat me about the head and shoulders again. :-)) The source of this info is the article "Communications Aspects of the Compact Disc Digital Audio System" from the Feb. 1985 issue of the IEEE Communications Magazine. The author, Dr. J.B.H. Peek is with Philips Research Labs. According to Dr. Peek, their interpolation filter is a 96-tap FIR filter. The input to the filter is 16-bit, and 3 zero samples are inserted between each input sample to increase the sample rate by a factor of 4. The filter has 12-bit coefficients, but the article has conflicting statements as to whether 24- or 28-bit arithmetic is used in the filter. A guess would be 24-bit because 12 by 12 multipliers are available (although a 16 by 16 could just as easily be used). For this length and complexity filter it is necessary to have a multiplier. Because of the inserted zero samples, only 24 multiplies are required per filter update. I assume there is also a 24-bit adder to round out the filter hardware. The filter passband is 0-22 kHz and the the stopband is 24-88.2 kHz. The article does not state what the passband ripple is, but the stopband attenuation is 50 dB. The output of the filter is rounded to 14 bits, which would have 6dB disadvantage compared to its 16-bit un-interpolated counterpart (interpolatation by 4 provides an effective 15-bit resolution). Then, there's a trick that Philips uses. They apply a noise shaping filter (no details provided) to the interpolator output. The noise shaping filter reduces the in-band (0-20kHz) noise at the expense of increased out-of-band noise. The article claims that this reduces the audible noise by another 7 dB thus making the interpolated system of comparible performance to the 16-bit system. As I stated in a previous posting, the interpolation process makes the analog anti-alias filter much simpler than the 44.1 kHz sample rate / 16-bit system. This, along with the replicability of digital filters from unit to unit argues in favor of the interpolative approach. The point I was trying to make with my original posting (I did a rather poor job) was that the nature of FIR filter implementation allows a reduction in the computational power required compared to its IIR counterpart. (Yes, I know the order of the IIR filter would be much lower than the FIR filter order.) The FIR filter allows an EFFECTIVE computational rate equal to the input sample rate because only 1 out every 4 multiplies needs to be performed. An equivalent IIR filter must be updated at the output sample rate due to the feedback involved. Thanks again for your patience. ---------- "...but you've got no arms and no legs, what are you going to do, bleed all over me ..." Steve Eidson (408) 749-2303 UUCP: {ucbvax,decwrl,ihnp4,allegra}!amdcad!amdimage!steve ARPA: amdcad!amdimage!steve@decwrl.ARPA